[Asterisk-Users] voice over atlantic

[asterisk groups]

David,
I have IAX trunks running between the US and S. America using the GSM
codec and jitterbuffer=yes and the quality seems very good to my ears.
Don't have the details of the jitterbuffer parameters right now but
hopefully this will give you some useful feedback.

Good luck.


On Thu, 2005-09-08 at 16:49 -0400, David Hajek wrote:
> Hi-
> 
> I'm using IAX between two boxes, where one box is located in US and the
> second in Europe. I'm trying to achieve the best voice quality and
> mainly reliability between these boxes and looking for hints and
> experience of others. 
> 
> Facts:
> - Asterisk 1.0.7
> - RTT varies from 130-170 ms, depends on time and actual Internet
> throughput
> 
> Questions:
> - What is the sugested codec for such setup? Now I'm using ULAW, but
> realizing it may not be the best choice. Sometimes I can hear broken
> audio. Maybe speex is better choice? 
> - Jitter buffer, yes/no? What are the suggested values. Currently I'm
> using these values:
> jitterbuffer=yes
> dropcount=10
> maxjitterbuffer=500
> maxexcessbuffer=300
> minexcessbuffer=20
> jittershrinkrate=2 
> - Trunking? Is it reliable enough?
>